audio: Enable 24 bit packed direct pcm support.
-Add support for 24 bit packed audio in audio hal.
-Disable gapless for AV playback and direct pcm usecase,
this ensures that the buffering in DSP is not more.
-Simulate rendered time stamp for direct pcm usecase
based on the number of frames written to the compress
driver, bufferring in the driver and DSP latency.
-Pass mixer instance to offload effects library to avoid
an unnecessary mixer_open call, this optimizes audio
start delay in compress playback.
Change-Id: I422a53af5632eaf6cc362a6c44f62ff8412965f7
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 91d7c0a..3ce83b1 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -40,17 +40,6 @@
#include <cutils/str_parms.h>
-#ifndef PCM_OFFLOAD_ENABLED
-#define AUDIO_FORMAT_PCM_OFFLOAD 0x1A000000UL
-#define AUDIO_FORMAT_PCM_16_BIT_OFFLOAD (AUDIO_FORMAT_PCM_OFFLOAD | AUDIO_FORMAT_PCM_SUB_16_BIT)
-#define AUDIO_FORMAT_PCM_24_BIT_OFFLOAD (AUDIO_FORMAT_PCM_OFFLOAD | AUDIO_FORMAT_PCM_SUB_8_24_BIT)
-#define AUDIO_OFFLOAD_CODEC_FORMAT "music_offload_codec_format"
-#define audio_is_offload_pcm(format) (0)
-#define OFFLOAD_USE_SMALL_BUFFER false
-#else
-#define OFFLOAD_USE_SMALL_BUFFER ((info->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM_OFFLOAD)
-#endif
-
#ifndef AFE_PROXY_ENABLED
#define AUDIO_DEVICE_OUT_PROXY 0x40000
#endif
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index b3ba2b5..cd9ead7 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -97,6 +97,8 @@
const struct string_to_enum s_format_name_to_enum_table[] = {
STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
STRING_TO_ENUM(AUDIO_FORMAT_MP3),
STRING_TO_ENUM(AUDIO_FORMAT_AAC),
@@ -117,8 +119,6 @@
STRING_TO_ENUM(AUDIO_FORMAT_QCELP),
STRING_TO_ENUM(AUDIO_FORMAT_MP2),
STRING_TO_ENUM(AUDIO_FORMAT_EVRCNW),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT_OFFLOAD),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_OFFLOAD),
STRING_TO_ENUM(AUDIO_FORMAT_FLAC),
STRING_TO_ENUM(AUDIO_FORMAT_ALAC),
STRING_TO_ENUM(AUDIO_FORMAT_APE),
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 8486e18..c53f124 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -74,6 +74,8 @@
#include "sound/asound.h"
#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
+/*DIRECT PCM has same buffer sizes as DEEP Buffer*/
+#define DIRECT_PCM_NUM_FRAGMENTS 2
/* ToDo: Check and update a proper value in msec */
#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
@@ -287,17 +289,17 @@
return ret_val;
}
-static int check_and_set_gapless_mode(struct audio_device *adev) {
-
-
- char value[PROPERTY_VALUE_MAX] = {0};
+static int check_and_set_gapless_mode(struct audio_device *adev, bool enable_gapless)
+{
bool gapless_enabled = false;
const char *mixer_ctl_name = "Compress Gapless Playback";
struct mixer_ctl *ctl;
ALOGV("%s:", __func__);
- property_get("audio.offload.gapless.enabled", value, NULL);
- gapless_enabled = atoi(value) || !strncmp("true", value, 4);
+ gapless_enabled = property_get_bool("audio.offload.gapless.enabled", false);
+
+ /*Disable gapless if its AV playback*/
+ gapless_enabled = gapless_enabled && enable_gapless;
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
@@ -323,8 +325,8 @@
format == AUDIO_FORMAT_AAC_ADTS_LC ||
format == AUDIO_FORMAT_AAC_ADTS_HE_V1 ||
format == AUDIO_FORMAT_AAC_ADTS_HE_V2 ||
- format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD ||
- format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD ||
+ format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
+ format == AUDIO_FORMAT_PCM_8_24_BIT ||
format == AUDIO_FORMAT_PCM_16_BIT ||
format == AUDIO_FORMAT_FLAC ||
format == AUDIO_FORMAT_ALAC ||
@@ -351,7 +353,6 @@
case AUDIO_FORMAT_AAC_ADTS:
id = SND_AUDIOCODEC_AAC;
break;
- case AUDIO_FORMAT_PCM_OFFLOAD:
case AUDIO_FORMAT_PCM:
id = SND_AUDIOCODEC_PCM;
break;
@@ -1956,7 +1957,7 @@
for the default max poll time (20s) in the event of an SSR.
Reduce the poll time to observe and deal with SSR faster.
*/
- if (out->use_small_bufs) {
+ if (!out->non_blocking) {
compress_set_max_poll_wait(out->compr, 1000);
}
@@ -1973,7 +1974,7 @@
if (adev->visualizer_start_output != NULL)
adev->visualizer_start_output(out->handle, out->pcm_device_id);
if (adev->offload_effects_start_output != NULL)
- adev->offload_effects_start_output(out->handle, out->pcm_device_id);
+ adev->offload_effects_start_output(out->handle, out->pcm_device_id, adev->mixer);
audio_extn_check_and_set_dts_hpx_state(adev);
}
}
@@ -2061,6 +2062,37 @@
return size;
}
+static uint64_t get_actual_pcm_frames_rendered(struct stream_out *out)
+{
+ uint64_t actual_frames_rendered = 0;
+ size_t kernel_buffer_size = out->compr_config.fragment_size * out->compr_config.fragments;
+
+ /* This adjustment accounts for buffering after app processor.
+ * It is based on estimated DSP latency per use case, rather than exact.
+ */
+ int64_t platform_latency = platform_render_latency(out->usecase) *
+ out->sample_rate / 1000000LL;
+
+ /* not querying actual state of buffering in kernel as it would involve an ioctl call
+ * which then needs protection, this causes delay in TS query for pcm_offload usecase
+ * hence only estimate.
+ */
+ int64_t signed_frames = out->written - kernel_buffer_size;
+
+ signed_frames = signed_frames / (audio_bytes_per_sample(out->format) * popcount(out->channel_mask)) - platform_latency;
+
+ if (signed_frames > 0)
+ actual_frames_rendered = signed_frames;
+
+ ALOGVV("%s signed frames %lld out_written %lld kernel_buffer_size %d"
+ "bytes/sample %zu channel count %d", __func__,(long long int)signed_frames,
+ (long long int)out->written, (int)kernel_buffer_size,
+ audio_bytes_per_sample(out->compr_config.codec->format),
+ popcount(out->channel_mask));
+
+ return actual_frames_rendered;
+}
+
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
@@ -2553,6 +2585,9 @@
out_standby(&out->stream.common);
return ret;
}
+ if ( ret == (ssize_t)bytes && !out->non_blocking)
+ out->written += bytes;
+
if (!out->playback_started && ret >= 0) {
compress_start(out->compr);
audio_extn_dts_eagle_fade(adev, true, out);
@@ -2627,14 +2662,24 @@
*dsp_frames = 0;
if (is_offload_usecase(out->usecase)) {
ssize_t ret = 0;
+
+ /* Below piece of code is not guarded against any lock beacuse audioFliner serializes
+ * this operation and adev_close_output_stream(where out gets reset).
+ */
+ if (!out->non_blocking && (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) {
+ *dsp_frames = get_actual_pcm_frames_rendered(out);
+ ALOGVV("dsp_frames %d sampleRate %d",(int)*dsp_frames,out->sample_rate);
+ return 0;
+ }
+
lock_output_stream(out);
- if (out->compr != NULL) {
+ if (out->compr != NULL && out->non_blocking) {
ret = compress_get_tstamp(out->compr, (unsigned long *)dsp_frames,
&out->sample_rate);
if (ret < 0)
ret = -errno;
ALOGVV("%s rendered frames %d sample_rate %d",
- __func__, *dsp_frames, out->sample_rate);
+ __func__, *dsp_frames, out->sample_rate);
}
pthread_mutex_unlock(&out->lock);
if (-ENETRESET == ret) {
@@ -2686,27 +2731,37 @@
int ret = -1;
unsigned long dsp_frames;
+ /* below piece of code is not guarded against any lock because audioFliner serializes
+ * this operation and adev_close_output_stream( where out gets reset).
+ */
+ if (is_offload_usecase(out->usecase) && !out->non_blocking &&
+ (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) {
+ *frames = get_actual_pcm_frames_rendered(out);
+ /* this is the best we can do */
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
+ ALOGVV("frames %lld playedat %lld",(long long int)*frames,
+ timestamp->tv_sec * 1000000LL + timestamp->tv_nsec / 1000);
+ return 0;
+ }
+
lock_output_stream(out);
- if (is_offload_usecase(out->usecase)) {
- if (out->compr != NULL) {
- ret = compress_get_tstamp(out->compr, &dsp_frames,
- &out->sample_rate);
- ALOGVV("%s rendered frames %ld sample_rate %d",
- __func__, dsp_frames, out->sample_rate);
- *frames = dsp_frames;
- if (ret < 0)
- ret = -errno;
- if (-ENETRESET == ret) {
- ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
- set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
- ret = -EINVAL;
- } else
- ret = 0;
-
- /* this is the best we can do */
- clock_gettime(CLOCK_MONOTONIC, timestamp);
- }
+ if (is_offload_usecase(out->usecase) && out->compr != NULL && out->non_blocking) {
+ ret = compress_get_tstamp(out->compr, &dsp_frames,
+ &out->sample_rate);
+ ALOGVV("%s rendered frames %ld sample_rate %d",
+ __func__, dsp_frames, out->sample_rate);
+ *frames = dsp_frames;
+ if (ret < 0)
+ ret = -errno;
+ if (-ENETRESET == ret) {
+ ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
+ set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
+ ret = -EINVAL;
+ } else
+ ret = 0;
+ /* this is the best we can do */
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
} else {
if (out->pcm) {
unsigned int avail;
@@ -2837,6 +2892,7 @@
ALOGD("copl(%p):calling compress flush", out);
lock_output_stream(out);
stop_compressed_output_l(out);
+ out->written = 0;
pthread_mutex_unlock(&out->lock);
ALOGD("copl(%p):out of compress flush", out);
return 0;
@@ -3238,7 +3294,6 @@
out->handle = handle;
out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
out->non_blocking = 0;
- out->use_small_bufs = false;
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL &&
(flags & AUDIO_OUTPUT_FLAG_DIRECT)) {
@@ -3337,6 +3392,9 @@
}
if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+ out->stream.pause = out_pause;
+ out->stream.flush = out_flush;
+ out->stream.resume = out_resume;
out->usecase = get_offload_usecase(adev, true);
ALOGV("DIRECT_PCM usecase ... usecase selected %d ", out->usecase);
} else {
@@ -3381,18 +3439,19 @@
out->compr_config.codec->id =
get_snd_codec_id(config->offload_info.format);
- if (((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM_OFFLOAD)||
- ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM)) {
+ if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
out->compr_config.fragment_size =
platform_get_pcm_offload_buffer_size(&config->offload_info);
+ out->compr_config.fragments = DIRECT_PCM_NUM_FRAGMENTS;
} else if (audio_extn_dolby_is_passthrough_stream(out)) {
out->compr_config.fragment_size =
audio_extn_dolby_get_passt_buffer_size(&config->offload_info);
+ out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
} else {
out->compr_config.fragment_size =
platform_get_compress_offload_buffer_size(&config->offload_info);
+ out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
}
- out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
out->compr_config.codec->sample_rate =
config->offload_info.sample_rate;
out->compr_config.codec->bit_rate =
@@ -3408,16 +3467,12 @@
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS)
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
- if (config->offload_info.format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD)
- out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
- if (config->offload_info.format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD)
- out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
if (config->offload_info.format == AUDIO_FORMAT_PCM_16_BIT)
out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
-
- if (out->bit_width == 24) {
+ if (config->offload_info.format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+ out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_3LE;
+ if (config->offload_info.format == AUDIO_FORMAT_PCM_8_24_BIT)
out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
- }
if (config->offload_info.format == AUDIO_FORMAT_FLAC)
out->compr_config.codec->options.flac_dec.sample_size = AUDIO_OUTPUT_BIT_WIDTH;
@@ -3425,14 +3480,6 @@
if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
out->non_blocking = 1;
- if (platform_use_small_buffer(&config->offload_info)) {
- //this flag is set from framework only if its for PCM formats
- //no need to check for PCM format again
- out->non_blocking = 0;
- out->use_small_bufs = true;
- ALOGI("Keep write blocking for small buff: non_blockling %d",
- out->non_blocking);
- }
out->send_new_metadata = 1;
out->send_next_track_params = false;
@@ -3446,11 +3493,18 @@
ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
__func__, config->offload_info.version,
config->offload_info.bit_rate);
- //Decide if we need to use gapless mode by default
- if (!audio_extn_dolby_is_passthrough_stream(out)) {
- ALOGV("%s: don't enable gapless for passthrough", __func__);
- check_and_set_gapless_mode(adev);
- }
+
+ /* Disable gapless if any of the following is true
+ * passthrough playback
+ * AV playback
+ * Direct PCM playback
+ */
+ if (audio_extn_dolby_is_passthrough_stream(out) ||
+ config->offload_info.has_video ||
+ out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+ check_and_set_gapless_mode(adev, false);
+ } else
+ check_and_set_gapless_mode(adev, true);
if (audio_extn_dolby_is_passthrough_stream(out)) {
out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
@@ -4288,7 +4342,7 @@
ALOGV("%s: DLOPEN successful for %s", __func__,
OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
adev->offload_effects_start_output =
- (int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
+ (int (*)(audio_io_handle_t, int, struct mixer *))dlsym(adev->offload_effects_lib,
"offload_effects_bundle_hal_start_output");
adev->offload_effects_stop_output =
(int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 31184d5..6c97840 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -209,7 +209,6 @@
struct stream_app_type_cfg app_type_cfg;
int non_blocking;
- bool use_small_bufs;
int playback_started;
int offload_state;
pthread_cond_t offload_cond;
@@ -343,7 +342,7 @@
int (*visualizer_start_output)(audio_io_handle_t, int);
int (*visualizer_stop_output)(audio_io_handle_t, int);
void *offload_effects_lib;
- int (*offload_effects_start_output)(audio_io_handle_t, int);
+ int (*offload_effects_start_output)(audio_io_handle_t, int, struct mixer *);
int (*offload_effects_stop_output)(audio_io_handle_t, int);
struct sound_card_status snd_card_status;
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 857d9e1..46611fc 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -760,6 +760,7 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
+#define PCM_OFFLOAD_PLATFORM_DELAY (30*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
static bool is_misc_usecase(audio_usecase_t usecase) {
@@ -3651,7 +3652,7 @@
free(kv_pairs);
}
-/* Delay in Us */
+/* Delay in Us, only to be used for PCM formats */
int64_t platform_render_latency(audio_usecase_t usecase)
{
switch (usecase) {
@@ -3659,6 +3660,9 @@
return DEEP_BUFFER_PLATFORM_DELAY;
case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
return LOW_LATENCY_PLATFORM_DELAY;
+ case USECASE_AUDIO_PLAYBACK_OFFLOAD:
+ case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
+ return PCM_OFFLOAD_PLATFORM_DELAY;
default:
return 0;
}
@@ -3846,19 +3850,17 @@
uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info)
{
uint32_t fragment_size = 0;
- uint32_t bits_per_sample = 16;
+ uint32_t bytes_per_sample;
uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION;
- if (info->format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD) {
- bits_per_sample = 32;
- }
+ bytes_per_sample = audio_bytes_per_sample(info->format);
//duration is set to 40 ms worth of stereo data at 48Khz
//with 16 bit per sample, modify this when the channel
//configuration is different
fragment_size = (pcm_offload_time
* info->sample_rate
- * (bits_per_sample >> 3)
+ * bytes_per_sample
* popcount(info->channel_mask))/1000;
if(fragment_size < MIN_PCM_OFFLOAD_FRAGMENT_SIZE)
fragment_size = MIN_PCM_OFFLOAD_FRAGMENT_SIZE;
@@ -3867,23 +3869,18 @@
// To have same PCM samples for all channels, the buffer size requires to
// be multiple of (number of channels * bytes per sample)
// For writes to succeed, the buffer must be written at address which is multiple of 32
- fragment_size = ALIGN(fragment_size, ((bits_per_sample >> 3)* popcount(info->channel_mask) * 32));
+ fragment_size = ALIGN(fragment_size, (bytes_per_sample * popcount(info->channel_mask) * 32));
ALOGI("PCM offload Fragment size to %d bytes", fragment_size);
return fragment_size;
}
-bool platform_use_small_buffer(audio_offload_info_t* info)
-{
- return OFFLOAD_USE_SMALL_BUFFER;
-}
-
/*
* configures afe with bit width and Sample Rate
*/
static int platform_set_codec_backend_cfg(struct audio_device* adev,
- snd_device_t snd_device,
- unsigned int bit_width, unsigned int sample_rate)
+ snd_device_t snd_device, unsigned int bit_width,
+ unsigned int sample_rate, audio_format_t format)
{
int ret = 0;
int backend_idx = DEFAULT_CODEC_BACKEND;
@@ -3908,13 +3905,17 @@
}
if (bit_width == 24) {
+ if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+ mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
+ else
mixer_ctl_set_enum_by_string(ctl, "S24_LE");
} else {
mixer_ctl_set_enum_by_string(ctl, "S16_LE");
}
my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
- ALOGD("%s:becf: afe: %s mixer set to %d bit", __func__,
- my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
+ ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl,
+ bit_width, format);
}
/*
@@ -4150,11 +4151,13 @@
int new_snd_devices[SND_DEVICE_OUT_END];
int i, num_devices = 1;
bool ret = false;
+ audio_format_t format;
backend_idx = platform_get_backend_index(snd_device);
new_bit_width = usecase->stream.out->bit_width;
new_sample_rate = usecase->stream.out->sample_rate;
+ format = usecase->stream.out->format;
ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
@@ -4171,7 +4174,7 @@
if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
&new_bit_width, &new_sample_rate)) {
platform_set_codec_backend_cfg(adev, new_snd_devices[i],
- new_bit_width, new_sample_rate);
+ new_bit_width, new_sample_rate, format);
ret = true;
}
}
@@ -4691,8 +4694,8 @@
format = DTS_HD;
break;
case AUDIO_FORMAT_PCM_16_BIT:
- case AUDIO_FORMAT_PCM_16_BIT_OFFLOAD:
- case AUDIO_FORMAT_PCM_24_BIT_OFFLOAD:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_8_24_BIT:
ALOGV("%s:PCM", __func__);
format = LPCM;
break;
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 5a41b07..2c60f3b 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1119,11 +1119,6 @@
return 0;
}
-bool platform_use_small_buffer(audio_offload_info_t* info)
-{
- return false;
-}
-
int platform_get_edid_info(void *platform __unused)
{
return -ENOSYS;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 385d20b..f1ec7a9 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -757,6 +757,7 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
+#define PCM_OFFLOAD_PLATFORM_DELAY (30*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
bool platform_send_gain_dep_cal(void *platform, int level) {
@@ -3677,7 +3678,7 @@
free(kv_pairs);
}
-/* Delay in Us */
+/* Delay in Us, only to be used for PCM formats */
int64_t platform_render_latency(audio_usecase_t usecase)
{
switch (usecase) {
@@ -3685,6 +3686,9 @@
return DEEP_BUFFER_PLATFORM_DELAY;
case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
return LOW_LATENCY_PLATFORM_DELAY;
+ case USECASE_AUDIO_PLAYBACK_OFFLOAD:
+ case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
+ return PCM_OFFLOAD_PLATFORM_DELAY;
default:
return 0;
}
@@ -3782,44 +3786,38 @@
uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info)
{
uint32_t fragment_size = 0;
- uint32_t bits_per_sample = 16;
+ uint32_t bytes_per_sample;
uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION;
- if (info->format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD) {
- bits_per_sample = 32;
- }
+ bytes_per_sample = audio_bytes_per_sample(info->format);
//duration is set to 40 ms worth of stereo data at 48Khz
//with 16 bit per sample, modify this when the channel
//configuration is different
fragment_size = (pcm_offload_time
* info->sample_rate
- * (bits_per_sample >> 3)
+ * bytes_per_sample
* popcount(info->channel_mask))/1000;
if(fragment_size < MIN_PCM_OFFLOAD_FRAGMENT_SIZE)
fragment_size = MIN_PCM_OFFLOAD_FRAGMENT_SIZE;
else if(fragment_size > MAX_PCM_OFFLOAD_FRAGMENT_SIZE)
fragment_size = MAX_PCM_OFFLOAD_FRAGMENT_SIZE;
+
// To have same PCM samples for all channels, the buffer size requires to
// be multiple of (number of channels * bytes per sample)
// For writes to succeed, the buffer must be written at address which is multiple of 32
- fragment_size = ALIGN(fragment_size, ((bits_per_sample >> 3)* popcount(info->channel_mask) * 32));
+ fragment_size = ALIGN(fragment_size, ((bytes_per_sample) * popcount(info->channel_mask) * 32));
ALOGI("PCM offload Fragment size to %d bytes", fragment_size);
return fragment_size;
}
-bool platform_use_small_buffer(audio_offload_info_t* info)
-{
- return OFFLOAD_USE_SMALL_BUFFER;
-}
-
/*
* configures afe with bit width and Sample Rate
*/
static int platform_set_codec_backend_cfg(struct audio_device* adev,
- snd_device_t snd_device,
- unsigned int bit_width, unsigned int sample_rate)
+ snd_device_t snd_device, unsigned int bit_width,
+ unsigned int sample_rate, audio_format_t format)
{
int ret = 0;
int backend_idx = DEFAULT_CODEC_BACKEND;
@@ -3845,13 +3843,16 @@
}
if (bit_width == 24) {
- mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+ if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+ mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
+ else
+ mixer_ctl_set_enum_by_string(ctl, "S24_LE");
} else {
mixer_ctl_set_enum_by_string(ctl, "S16_LE");
}
my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
- ALOGD("%s:becf: afe: %s mixer set to %d bit", __func__,
- my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
+ ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
}
/*
@@ -4053,11 +4054,13 @@
int i, num_devices = 1;
bool ret = false;
struct platform_data *my_data = (struct platform_data *)adev->platform;
+ audio_format_t format;
backend_idx = platform_get_backend_index(snd_device);
new_bit_width = usecase->stream.out->bit_width;
new_sample_rate = usecase->stream.out->sample_rate;
+ format = usecase->stream.out->format;
ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
@@ -4073,7 +4076,7 @@
if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
&new_bit_width, &new_sample_rate)) {
platform_set_codec_backend_cfg(adev, new_snd_devices[i],
- new_bit_width, new_sample_rate);
+ new_bit_width, new_sample_rate, format);
ret = true;
}
}
@@ -4591,8 +4594,8 @@
format = DTS_HD;
break;
case AUDIO_FORMAT_PCM_16_BIT:
- case AUDIO_FORMAT_PCM_16_BIT_OFFLOAD:
- case AUDIO_FORMAT_PCM_24_BIT_OFFLOAD:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_8_24_BIT:
ALOGV("%s:PCM", __func__);
format = LPCM;
break;
diff --git a/hal/platform_api.h b/hal/platform_api.h
index df80a0c..cb177b6 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -109,7 +109,6 @@
struct audio_offload_info_t;
uint32_t platform_get_compress_offload_buffer_size(audio_offload_info_t* info);
uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info);
-bool platform_use_small_buffer(audio_offload_info_t* info);
uint32_t platform_get_compress_passthrough_buffer_size(audio_offload_info_t* info);
bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index d39a8b7..464bc0d 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -209,7 +209,7 @@
* Interface from audio HAL
*/
__attribute__ ((visibility ("default")))
-int offload_effects_bundle_hal_start_output(audio_io_handle_t output, int pcm_id)
+int offload_effects_bundle_hal_start_output(audio_io_handle_t output, int pcm_id, struct mixer *mixer)
{
int ret = 0;
struct listnode *node;
@@ -245,19 +245,19 @@
/* populate the mixer control to send offload parameters */
snprintf(mixer_string, sizeof(mixer_string),
"%s %d", "Audio Effects Config", out_ctxt->pcm_device_id);
- out_ctxt->mixer = mixer_open(MIXER_CARD);
- if (!out_ctxt->mixer) {
- ALOGE("Failed to open mixer");
+
+ if (!mixer) {
+ ALOGE("Invalid mixer");
out_ctxt->ctl = NULL;
out_ctxt->ref_ctl = NULL;
ret = -EINVAL;
free(out_ctxt);
goto exit;
} else {
+ out_ctxt->mixer = mixer;
out_ctxt->ctl = mixer_get_ctl_by_name(out_ctxt->mixer, mixer_string);
if (!out_ctxt->ctl) {
ALOGE("mixer_get_ctl_by_name failed");
- mixer_close(out_ctxt->mixer);
out_ctxt->mixer = NULL;
ret = -EINVAL;
free(out_ctxt);
@@ -314,9 +314,6 @@
fx_ctxt->ops.stop(fx_ctxt, out_ctxt);
}
- if (out_ctxt->mixer)
- mixer_close(out_ctxt->mixer);
-
list_remove(&out_ctxt->outputs_list_node);
#ifdef DTS_EAGLE